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One of the main functions of rostrvm is to provide users and agents access to a telephony switch. Normally this will include some type of a Voice over IP (VoIP) gateway positioned between your particular telephony switch or telephony infrastructure and your rostrvm system. This type of architecture is known as an Invex system and is the configuration documented here (for non-Invex systems please contact Rostrvm Solutions Ltd).
Invex is a specific type of rostrvm system that uses Voice over IP to enhance the functionality provided by a less sophisticated PBX device, or VoIP telephones that support the SIP protocol. In the former more common case, communications with the switch are normally through a VoIP telephony gateway such as a Vega or Cisco.
Dial Plan Tab
This is information is required to make outbound calls via one of more routers (e.g. Cisco or Vega) used to connect to legacy telephony devices (such as subscriber phones within the PSTN). The details should be obtained when the gateway is installed.
The destination for a telephone call is matched against the list of Dial Plans starting at the top and working down (note a blank dial plan matches all destinations). Once a match is found then that router is used to make the call. Multiple entries with the same dial plan can be defined. If the call fails on the first one (e.g. all trunks are busy) then it is made on the next match in the list.
To clarify through an example, consider the following configuration:
9* 192.168.111.88 5060 0
8* 192.168.111.99 5060 0
192.168.111.55 5060 0
192.168.111.44 5060 0
With the Number Of Ports all set to 0, any outbound call that does not begin with a '9' or an '8' will be attempted on 192.168.111.55 first. If all the trunks are busy (or it fails for any other reason) then 192.168.111.44 will be tried instead. All calls that begin with a '9' will try 192.168.111.88 first, but will then try 55 and 44 if this fails.
However, if the number of ports is set as follows:
9* 192.168.111.88 5060 0
8* 192.168.111.99 5060 0
192.168.111.55 5060 10
192.168.111.44 5060 20
Load sharing is invoked. In this case all outbound calls (regardless of the first digit) will be routed through gateway 192.168.111.44 (as this has the most ports available). Gateway 192.168.111.55 will only be used once 44 has at least 10 calls in progress. If this is the case then any new outbound calls will be shared between 44 and 55. Gateways 88 and 99 will only be used if the dial plan matches AND gateway 44 has at least 20 calls, and gateway 55 has at least 10 calls.
The number of ports is not taken literally by rostrvm, it is the difference between entries that is important. So in the above example, attempts to make calls can still be made even when gateway 44 has more than 20 calls in progress, and gateway 55 has more than 10 calls in progress.
By default a dial plan is configured for 901632* and 01632* (fictional STD codes) to connect to the VoIP Contact Simulator.
| • | Dial Plan - Specifies the outbound dial plan applicable to the PSTN gateway routers; the order defines their priority. The order of the entries can be changed by click and dragging an entry to the required position. |
| • | Host / Port - This should be set to the IP address and port identifying where to route calls defined by the Dial Plan. This is normally a Gateway Router but can also be another SIP Proxy. |
| • | Origin - The default number to be used as the presentation CLI for remote calls over the SIP Trunk / gateway. This can be overridden by the number configured against specific Call Classes; |
| • | Number of Ports - This will normally be set to 0 unless load-sharing is required. If set to anything other than 0 then if the dial plan matches multiple entries then the one with the most ports remaining will be used in preference to the other. Warning: if only 1 entry has Number of Ports set then load-sharing will be invoked between ALL matching dial plans even if the others that match are set to 0. |
From here the user can:
| • | By clicking the Add Row button create a new dial plan entry; |
| • | By clicking the Delete button on the right delete the dial plan entry. |
Gateway Status
The status of each configured gateway is checked every Request Timeout (default 10 seconds). If it responds before the next ping then the gateway is marked online. If it does not respond before the next ping then it will be marked as offline and an alarm raised. Offline gateways are ignored and excluded from the dial plan and load-sharing logic described above.
Number Manipulation Tab
When making an outgoing call the destination telephone number provided may be manipulated in a number of ways.
| • | International Prefix - this mapping is applied to all rostrvm initiated outgoing calls (i.e. make call, make predictive call, consultation call, requeue call, or route select). It determines what any leading ‘+’ character is replaced with for any international telephone numbers being dialled. This is optional and defaults to '00' if not configured; |
| • | Outbound Number Prefix - this mapping is applied to all rostrvm initiated outgoing calls (i.e. make call, make predictive call, consultation call, requeue call, or route select). It is added to the beginning of the telephone number before it is dialled; |
| • | Prefix / Substitute - the entries in this table are applied to remote telephone numbers for both inbound and outbound calls but for MIS recording purposes only (i.e. these changes are not applied to real-time outbound make call requests). They are however also applied to the account telephone numbers as part of the de-duplication logic during list import. |
Note that for dialler initiated calls it is also possible to add a prefix to each telephone number on a campaign by campaign basis. This is configured under the campaign properties page.
From here the user can:
| • | By clicking the Add Prefix button create a new mapping entry; |
| • | Delete an entry by clicking the Delete button on the right of the prefix mapping row. |
Tones Tab
From here you can set a variety of tones and music to be played to the customer on receipt of an inbound call or initiation of an outbound call.
| • | Ring Tone - a drop-down list of tones that can be played when calling a remote and the remote is alerting (and no voice path is available to the remote); |
| • | Dialling Tone - a drop-down list of tones that can be played when calling a remote and the remote state is unknown (and no voice path is available to the remote); |
| • | Incoming Tone - a drop-down list of tones that can be played when an incoming call is presented to the device (for ScreenPhone this is used for auto-answer calls); |
| • | Incoming Message Tone - a drop-down list of tones that can be played for incoming messages (i.e. emails and web chats); |
| • | ScreenPhone Incoming Ring - a drop-down list of tones that can be played when an incoming call is presented to a ScreenPhone controlled device (except for auto-answer calls, see Incoming Tone); |
| • | Comfort Beep Interval - the interval in seconds between comfort beeps - if this is zero then no comfort beeps are played. The beep played is named ComfortBeep within the AutoAgent announcements tool; |
| • | Queue / Hold Music - these music loops reflect '.wav' files stored in the rostrvm database available for use by Invex components for playing in queue music and hold music respectively. |
The drop-down list display for the Tone fields are available in the rostrvm database which can be created from the Announcementsreport. Similarly the queue and hold music files are also imported into the rostrvm database using this utility. It is important that new '.wav' files are imported this way (rather than other means such as File Editor) as it ensures that the files are stored in the correct format.
From here the user can:
| • | Navigate to the the announcement report by clicking the Announcements button. |
Text to Speech Tab
From here you can configure which text to speech voice is used.
| • | Text to Speech Voice - choose from the drop-down menu which text to speech voice you wish to be used (David, Hazel, Zira); |
| • | Sample - type in the text to speak and then click the Play button to hear the audio. |
Informal Call Centre Tab
This allows remote agents to access limited rostrvm functions without the use of a PC. Agent control is achieved using DTMF over the agent's phone. For more details of the options available in an informal contact centre please contact Rostrvm Solutions Ltd.
| • | Confirm Logon Announcement - if set, this announcement will be played to an agent that has just logged on and been connected through an outbound nail up. The agent will be requested to confirm the connection through DTMF. |
| • | Confirm Logoff Announcement - if set, agents logging on to an Agent Phone with DTMF will be requested for a log off time – at this log off time this announcement will be played to the agent. |
| • | No Contact Action - the action to take after a call that did not result in an established connection. This has one of the following values: - Transfer to handler (this option is only available if a handler has been set); - No action; - Set agent ready. |
| • | Logoff Timeout (seconds) - the time before an agent in wrap-up should be logged off if the agent caller disconnects. This defaults to 1 minute. |
| • | Enable Keypad Control - a flag to indicate that agents can log in and control their state using their telephone keypad. This should only be set if you are not using AdVisor (or similar) screen based call control. |
Nail Up Tab
When agents logon to an Invex system, a telephone call is made to their turret/phone so that they become 'nailed-up' to the rostrvm system. Alternatively an agent can dial into the rostrvm system from their phone to become 'nailed-up'. For this latter method a telephone number needs to be configured on the gateway so as to route these agent incoming calls to the IP address of the rostrvm system and port as configured in SIP Port. Both methods require a number of rostrvm parameters to be configured.
| • | Nail Up Presentation - the CLI presented to the agent's turret when the nail-up call is delivered and established. For an inbound nail-up this should be the telephone number that the agent dials in to the rostrvm server; |
| • | Dialling Prefix - the prefix to be added when initiating the outbound nail-up call; |
| • | SIP Port - the SIP protocol port on which to listen for SIP connections. This defaults to 5066 and can be a number in the range 1024 to 65535. For multi-domain systems this needs to be unique for each multi-domain server. This is the port number that should be configured on the gateway to enable agent incoming nail-ups; |
| • | Disconnect After Each Call - a flag to indicate that nail up links should be cleared and re-established between remote calls (for external voice recorders); |
| • | Nail Up Timeout - the time in seconds that the system will wait for a nail-up call to be answered (default is 0 seconds). If the call has not been answered in this time then the attempt is cleared and the agent is logged off. |
From here the user can:
| • | Create a new entry by clicking the New Nail Up Presentation button. |
Contact Handling Tab
Allows the setting of some agent contact handling defaults for the entire rostrvm system. These can be overridden on a per user or per agent group basis.
| • | Wrap Up Timeout - this specifies the time an agent will spend in wrap up at the end of a call to be specified. The timeout type can be: Infinite, 1, 2, 3, 4 or 5 minutes or you can specify a number of seconds; |
| • | Auto Answer ACD Calls / Auto Answer Non ACD Calls - this specifies if rostrvm will provide auto-answer for calls received at the switch. It can be configured to either No, Allow Agent to Set (ACD only), Immediate, or a specified number of seconds (where 0 is the same as immediate). The auto answer can be set differently for ACD delivered calls and non-ACD delivered calls; |
| • | Busy / Not Ready / Wrap Up Routes - this specifies an alternative route if a direct call is made to agent in any of these states. Note that this does not apply to ACD calls, these are only presented to Ready agents; |
| • | Maximum Voice / Email / Chat Contacts - sets the number of voice, email and chat contacts that can be handled simultaneously. For voice this is either 0 (can't handle voice contacts) or 1 (the default). For email and chats this can be any number (although a sensible limit is advised); |
| • | Voice / Email / Chat Contact Rules - determines whether each type of contact can interrupt any, one or all of the other types of contact. It is normal to allow emails to interrupt voice and chat, and to allow voice and chat to interrupt emails only. It is not recommended to allow voice calls to interrupt chats and vice versa. |
| • | Agent Idle Timeout - the number of seconds that an agent can remain in the Ready state until an agent Idle event is generated over the HTTP Restful API event stream (see USM/002 rostrvm InterFace Reference Guide for details). |
Gateway Tab
rostrvm supports a variety of different types and versions of VoIP gateways. This tab allows some of the specific gateway configuration settings to be set and adjusted. Changes to these values should only be undertaken in consultation with Rostrvm Solutions Ltd.
| • | STUN Server/External IP address - this may be set to an external/public IP address that the rostrvm system is seen as on the internet. It may also to set to use STUN to determine the internet address for media. The parameter may specify stun: followed by the name of a stun server. If blank a STUN request is made to stun.xxxx.xxx (where xxxx.xxx is the parent domain of the system e.g. stun.rostrvm.local) and to a list of public STUN services. Normally this should be left blank. It should only be set where there is some special private networking/VPN/WAN involved. |
| • | SIP Port (Incoming Calls) - defines the SIP TCP port used for incoming calls - this is the port that the gateway should be configured to which to send calls. This setting should normally be left blank for automatic selection (5060 for a single domain rostrvm system). A firewall opening for this port should be opened to allow ScreenPhone to work over the internet. |
| • | Hold Type - the SIP standard defines two different methods for putting calls on hold, by default rostrvm uses both methods. Some Cisco configurations prefer the RFC3261 method to be used. Some older systems may prefer the RFC2543 method. |
| • | Link Protocol - set the protocol used for SIP messages - some older hardware requires UDP, some newer systems require TCP. Care should be taken a changing this setting may stop calls from working; |
| • | Voice Codec - this setting may be used to force the use of a particular voice codec in SIP calls. Normally this should be set to ALL to allow the system to determine the best codec; |
| • | Maximum Calls Per Second - the maximum number of calls that will be launched by rostrvm every second, 0 disables this feature. This can be used to throttle outgoing calls where there are hardware limits on call rates; |
| • | Register routing numbers with gateways - if set SIP REGISTER requests will be sent to each configured gateway for each configured routing number. This allows automatic configuration of gateways for incoming numbers in resilient systems. |
| • | Register stations with gateways - if set SIP REGISTER requests will be sent to each configured gateway for each configured station. This allows automatic configuration of gateways for incoming numbers direct to stations in resilient systems |
Advanced Tab
From here you can adjust your system performance by changing various timeouts and limits to suit your needs. Care should be taken when setting these parameters as it may affect your system in a negative rather than a positive way.
| • | Request Timeout - the time-out value before a request to the switch is assumed to have failed in seconds. It is recommended that this value be set to 0 so that the default (10 seconds) is used; |
| • | Predictive Request Timeout - the time-out value before a make predictive call request to the switch is removed from the queue. On some networks where mobile phones are being called, the time to reach the remote telephone can be quite a long time. It is recommended that this value be set to 0 so that the default (45 seconds) is used; |
| • | Number of Unique Digits - the number of unique digits on the end of a telephone number used to identify a remote. This is to avoid problems with prefixes that are added to numbers by some gateways; |
| • | Do not log off agent when the AdVisor disconnects - indicates if agents should be logged off if connection is lost to the AdVisor application. This setting should normally not be checked unless your system suffers from unexpected network connection problems; |
| • | Auto Logoff Time - time of day, in 24-hour clock format, at which all agents will be automatically logged off. This may be used to log off all agents at 11pm, for example, if agents leave their stations logged on at the end of a shift. Setting this to blank disables this feature. |
Operations
From here you can:
| • | Save all changes by clicking the Save button. |